vicidial.org

by **duodom** » Thu Apr 21, 2011 9:57 pm

I have installed the Vicibox Redux 3.1.9 x64 from ISO. Installed the latest dahdi tools as per Sangoma driver installation. Not able to dial out at all. All the info I found refers to SIP or IAX outgoing calls. Newbie to linux and dialers. Read the manual a few times. but there are no specifications to my problem. I am trying to deploy it as an predictive dialer. please help!

by **mflorell** » Fri Apr 22, 2011 6:26 am

ViciDial does not work well with Analog POTS lines because of the horrible call progress analysis in Asterisk. You would be better off using IAX, SIP or some form of T1/E1 connection.

by **williamconley** » Fri Apr 22, 2011 8:32 am

If you installed and then tried to set up according to the Vicidial Managers Manual, you will ultimately be able to use the system. But where did you bump into your first problem?

You said:

but there are no specifications to my problem

without mentioning your problem that has no "specifications". This is the rough equivalent of complaining that your car "don't work" and asking for help fixing it. If the description is "don't work", we will likely need a bit more information to help you fix it, yes?

So ... let's start with where you are in the manual ... what page? what line? (where you bump into your first problem ... starting at the beginning of the manual without skipping anything!)

And mflorell is right, you should at least add a single SIP line (to get the system up and tested, and learn what it can do) and then we can configure any other trunks you may want to use after we know your system is installed and working.

For the future, remember that when asking for free technical help, it is best to provide technical details.

by **striker** » Fri Apr 22, 2011 12:40 pm

have you confugured your sangoma card properly.

type

wancfg_dahdi this will lead you to the sangoma card configuration.

afer finishing the above steps type dahdi_cfg -vvvvv

this will outputs your channels

by **duodom** » Fri Apr 22, 2011 1:23 pm

the card is setup properly and it shows 16 live channels. Did the wancfg_dahdi and dahdi_cfg -vvvvv Everything seems to be fine. I would post the output but don't know how. Regarding previos posts. Following the manual worked as clock work. the time I noticed i have a problem was when i setup a campaing and no calls came through. watching the screen output using asterisk -r the meesage displayed when attempting to call was:
chan_local.c:599 local_call No such extension/context 4167254428@default while calling Local channel
channel.c: 3612 _ast_request_and_dial: unable to call channel Local/4167254428@default

by **williamconley** » Fri Apr 22, 2011 2:19 pm

you'll need to modify your extensions.conf or create a carrier to send the calls to your dahdi channels (or you can try dialing 9 o 91 before each phone number with a dial prefix in the campaign).

the system is saying it has no dial pattern matching that 10 digit phone number ... so the call fails. This is usually provided by the dial plan area of the carrier (but since your carrier isn't SIP or IAX, you did not need to add a carrier ... except now you need a dialplan to get the call to go to the dahdi channels).

have you dialed manually on this card successfully?

by **duodom** » Fri Apr 22, 2011 2:33 pm

I don't know how to dial by command line. I make the recommanded change to the extensions.conf. I am wondering if TRUNK=Zap/r1 and TRUNK=Zap/r2 are the problem. I have created a carrier in the admin portion of the Vicidial manager. But there where no changes. how can I create the dialplan. thank you for taking the time and helping me with this. I feel like a complete idiot at this time. also for the chan_dahdi.conf I have group=0.

by **williamconley** » Fri Apr 22, 2011 2:36 pm

i was not talking about dialing by command line, but manually with your sip phone.

show your dial plan that you entered

also, since you are set up as group=0 try changing Zap/r1 to Zap/g0, then use TRUNK as your variable in your dial plan.

by **duodom** » Fri Apr 22, 2011 2:49 pm

exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,2,Dial(${TRUNK}/${EXTEN:1},,To)
exten => _91NXXNXXXXXX,3,Hangup
as per manual.

I can see at least one of my issues here :oops:

I will add one more dial plan for local calling that include the area code. maybe will fix it. An I will change the name. I will post back in a few min. with the results.

by **duodom** » Fri Apr 22, 2011 3:11 pm

exten => _9NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9NXXNXXXXXX,2,Dial(${TRUNK}/${EXTEN:1},,To)
exten => _9NXXNXXXXXX,3,Hangup

added this in dialplan for local calling, did a reboot, tried a manual call and the auto-dialing now I also have a warning message that i have not seen before: file.c:1297 waitstream_core: unexpecte control subclass "-1"

by **duodom** » Fri Apr 22, 2011 3:35 pm

Is the 9 in:
exten => _9NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9NXXNXXXXXX,2,Dial(${TRUNK}/${EXTEN:1},,To)
exten => _9NXXNXXXXXX,3,Hangup
necessary?
I am dialing straigt 10 digit phone number.

by **williamconley** » Fri Apr 22, 2011 3:39 pm

note the "9" at the beginning of the dial pattern. when manual dialing, you must add the 9. when dialing in a campaign, the campaign dial prefix should supply the 9.

also of note is the need (or lack thereof?) to dial a 1 based on where you are dialing and whether it is in the dialplan and whether you loaded in a "1" as the dial code in the campaign.

first things first: manual call ... dial 9+10digit will match 9NXXNXXXXXX and should go through "Trunk". and be visible in the CLI then going out through ZAP/g0/9??????????

by **duodom** » Fri Apr 22, 2011 3:51 pm

Part of the problem was the 9. I don't have to dial 9 to access a phone line. but now I have a different warning
channel.c:3765 ast_request: Nochannel type registered for 'Zap'
app_dial.c :1310 dial_exec_full: Unable to create chane\nel of type "Zap' (cause 66 - channel not implemented).

Should I define the TRUNK as DAHDI?

by **duodom** » Fri Apr 22, 2011 4:10 pm

changed TRUNK=Zap/g0 to TRUNK=DAHDI/g0
everything seems to be working now. All I have to figure out is why it tells me the number canot be completed as dialed and that is something that i probably screwed up in loading the data. Thank you very much for your help. I really appreciated this. If you are ever in toronto I owe you at least a beer.

by **williamconley** » Fri Apr 22, 2011 5:46 pm

Sounds like a plan. Make it Pizza and Pepsi, though. I hate beer. LOL

You'll likely need to institute standard US 10 digit dialing, which requires 11 digits!

If you must dial 1+10digits to dial long distance, then 9NXXNXXXXXX won't do it ... it should be 91NXXNXXXXXX and you should have a "dial code" of 1 on each number in your database. If you missed the 1 when loading leads (which should not be possible, by the way), you CAN put "91" for the dial prefix on the campaign this time until you re-load your leads.

by **striker** » Fri Apr 22, 2011 7:37 pm

try this
in general
TRUNKXYZ=DAHDI/g0

in default context

exten => 4167254428,1,AGI(agi://127.0.0.1:4577/call_log)
exten => 4167254428,2,Dial(${TRUNKXYZ}/${EXTEN},,To)
exten => 4167254428,3,Hangup

now dial 4167254428 from any extensions.
as i have seen ur post you are dialling 4167254428 ,so
make sure 4167254428 can be dialled form any of your analog line using a analog phone.

if the call doesnt connects , post the cli result here

by **duodom** » Fri Apr 22, 2011 9:14 pm

Thank you very much guys. Works amazing. apparently my phone line provider requires a 1 in front of all numbers (lo0cal and long distance). Love the way this dialer works. again many thanks.
by the way Pizza and Pepsi it is LOL

by **williamconley** » Fri Apr 22, 2011 9:45 pm

Yeah, but you gotta get striker a beer or he'll probably pout.

by **duodom** » Fri Apr 22, 2011 9:49 pm

Not an issue. Pizza , Pepsi and Beer

by **raghurambharatham** » Tue May 03, 2011 7:51 am

hi:
good evening.i have installed goautodial and i like to call to my number something like +919xxxxxxxxx.how can i call.i have placed analog card in my server.what the commands that to follow.how can i do outbound calls through vicidial/goautodial plz tell me.

by **williamconley** » Tue May 03, 2011 2:34 pm

1)

when you post, please post your entire configuration including (but not limited to) your installation method and vicidial version with build.

this IS a requirement for posting along with reading the stickies (at the top of each forum) and the manager's manual (available on EFLO.net, both free and paid versions)

You should also post: Asterisk version, telephony hardware (model number is helpful here), cluster information if you have one, and whether any other software is installed in the box. If your installation method is "from scratch" you must post your operating system and should also post the .iso version from which you installed your original operating system.

Similar to This:

Vicibox X.X from .iso | Vicidial X.X.X-XXX Build XXXXXX-XXXX | Asterisk X.X.X | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation

2)

Show your present working dialplan for dialing another area, we'll help you convert it to this pattern

by **will_brook** » Fri Aug 05, 2011 8:47 am

Hi,

I have the following setup

Vicibox_redux.i686-3.1.10 from .iso Asterisk 1.4.39.2-vici | Single Server | Sangoma A104 hardware, 4 x E1 pstn lines | No Extra Software After Installation.

I'm pretty new to Vicidial and looking to get a test platform setup. I've installed off the CD and connecting my Sangoma card to two of the four E1 ports available but I'm unable to make any external calls from an X-lite phone.

I've run wancfg_dahdi, and setup all ports. I've also checked dahdi-channels.conf and can see group 0 setup. So I've changed the extension.conf file to the following:

TRUNK=DAHDI/g0
TRUNKX=DAHDI/g0

exten => _9XXXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9XXXXXXXXXXX,2,Dial(${TRUNKX}/${EXTEN},,To)
exten => _9XXXXXXXXXXX,3,Hangup

I'm based in the UK, so we have 11 digit numbers. I've also tried the options TRUNK=Zap/g0 & removed the 9 from the exten. Also added {EXTEN:1} as decribed in the manager manual.

When I try to run a test call from X-lite I get this error on asterisk -r. I've replaced my number with 907XXXXXXXXX.

[Aug 5 14:41:43] -- Executing [907XXXXXXXXX@default:1] AGI("SIP/101-00000000", "agi://127.0.0.1:4577/call_log") in new stack
[Aug 5 14:41:43] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Aug 5 14:41:43] -- Executing [907XXXXXXXXX@default:2] Dial("SIP/101-00000000", "DAHDI/g0/907XXXXXXXXX||To") in new stack
[Aug 5 14:41:43] -- Requested transfer capability: 0x00 - SPEECH
[Aug 5 14:41:43] -- Called g0/907XXXXXXXXX
[Aug 5 14:41:43] -- Channel 0/1, span 1 got hangup, cause 28
[Aug 5 14:41:43] -- Hungup 'DAHDI/1-1'
[Aug 5 14:41:43] == Everyone is busy/congested at this time (1:0/0/1)
[Aug 5 14:41:43] -- Executing [907XXXXXXXXX@default:3] Hangup("SIP/101-00000000", "") in new stack
[Aug 5 14:41:43] == Spawn extension (default, 907XXXXXXXXX, 3) exited non-zero on 'SIP/101-00000000'
[Aug 5 14:41:43] -- Executing [h@default:1] DeadAGI("SIP/101-00000000", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----28-----CHANUNAVAIL----------") in new stack
[Aug 5 14:41:43] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0

Any ideas I've reached a dead end and need your help.

by **will_brook** » Fri Aug 05, 2011 8:56 am

I forgot to mention that I have tested the two channels I have available via another PBX and they do work.

I've also run some wanrouter commands and can see the two lines are active:

Wanpipe Config:

Device name | Protocol Map | Adapter | IRQ | Slot/IO | If's | CLK | Baud rate |
wanpipe1 | N/A | A101/1D/A102/2D/4/4D/8| 28 | 1 | 1 | N/A | 0 |
wanpipe2 | N/A | A101/1D/A102/2D/4/4D/8| 28 | 1 | 1 | N/A | 0 |
wanpipe3 | N/A | A101/1D/A102/2D/4/4D/8| 28 | 1 | 1 | N/A | 0 |
wanpipe4 | N/A | A101/1D/A102/2D/4/4D/8| 28 | 1 | 1 | N/A | 0 |

Wanrouter Status:

Device name | Protocol | Station | Status |
wanpipe1 | AFT TE1 | N/A | Connected |
wanpipe2 | AFT TE1 | N/A | Connected |
wanpipe3 | AFT TE1 | N/A | Disconnected |
wanpipe4 | AFT TE1 | N/A | Disconnected |

wanrouter summary

Configuration File Summary in : /etc/wanpipe/

Device Protocol Type Cpu/Io Slot/Irq Bus State
------------------------------------------------------------------------
wanpipe1 WAN_AFT_TE1 PCI A 1 1 Connected
wanpipe2 WAN_AFT_TE1 PCI A 1 1 Connected
wanpipe3 WAN_AFT_TE1 PCI A 1 1 Disconnected
wanpipe4 WAN_AFT_TE1 PCI A 1 1 Disconnected
TESTPBX01:~ # wanrouter status

Devices currently active:
wanpipe1 wanpipe2 wanpipe3 wanpipe4

by **williamconley** » Sat Aug 06, 2011 10:53 am

exten => _9XXXXXXXXXXX,2,Dial(${TRUNKX}/${EXTEN},,To)

change to (Note the :1 at the end of EXTEN:1)

exten => _9XXXXXXXXXXX,2,Dial(${TRUNKX}/${EXTEN:1},,To)

because

Aug 5 14:41:43] -- Executing [907XXXXXXXXX@default:2] Dial("SIP/101-00000000", "DAHDI/g0/907XXXXXXXXX||To") in new stack

should be

Aug 5 14:41:43] -- Executing [907XXXXXXXXX@default:2] Dial("SIP/101-00000000", "DAHDI/g0/07XXXXXXXXX||To") in new stack

(note the missing "9" from the original "907")

by **will_brook** » Mon Aug 08, 2011 4:09 am

Hi,

Thanks for the quick response. I've applied your recommended changes and rebooted the server. However it's still not adding the 9 when I make an external call. See asterisk -r log:

[Aug 8 09:56:53] -- Executing [90XXXXXXXXXX@default:1] AGI("SIP/101-00000000", "agi://127.0.0.1:4577/call_log") in new stack
[Aug 8 09:56:53] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Aug 8 09:56:53] -- Executing [90XXXXXXXXXX@default:2] Dial("SIP/101-00000000", "DAHDI/g0/0XXXXXXXXXX||To") in new stack
[Aug 8 09:56:53] -- Requested transfer capability: 0x00 - SPEECH
[Aug 8 09:56:53] -- Called g0/07843036503
[Aug 8 09:56:54] -- Channel 0/1, span 1 got hangup, cause 28
[Aug 8 09:56:54] -- Hungup 'DAHDI/1-1'
[Aug 8 09:56:54] == Everyone is busy/congested at this time (1:0/0/1)
[Aug 8 09:56:54] -- Executing [90XXXXXXXXXX@default:3] Hangup("SIP/101-00000000", "") in new stack
[Aug 8 09:56:54] == Spawn extension (default, 90XXXXXXXXXX, 3) exited non-zero on 'SIP/101-00000000'
[Aug 8 09:56:54] -- Executing [h@default:1] DeadAGI("SIP/101-00000000", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----28-----CHANUNAVAIL----------") in new stack
[Aug 8 09:56:54] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0

by **will_brook** » Mon Aug 08, 2011 9:00 am

My mistake you said it should remove the 9, which it has. However the external call still doesn't connect.

Can you notice any other issues will the call log?

I've attached the settings from chan_dahdi.conf, is there anything wrong within this setup?

[trunkgroups]

[channels]
context=default
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no

;Sangoma A104 port 1 [slot:1 bus:1 span:1] <wanpipe1>
switchtype=euroisdn
context=from-pstn
group=0
echocancel=yes
signalling=pri_cpe
channel =>1-15,17-31

;Sangoma A104 port 2 [slot:1 bus:1 span:2] <wanpipe2>
switchtype=euroisdn
context=from-pstn
group=0
echocancel=yes
signalling=pri_cpe
channel =>32-46,48-62

;Sangoma A104 port 3 [slot:1 bus:1 span:3] <wanpipe3>
switchtype=euroisdn
context=from-pstn
group=0
echocancel=yes
signalling=pri_cpe
channel =>63-77,79-93

;Sangoma A104 port 4 [slot:1 bus:1 span:4] <wanpipe4>
switchtype=euroisdn
context=from-pstn
group=0
echocancel=yes
signalling=pri_cpe
channel =>94-108,110-124

by **will_brook** » Thu Aug 18, 2011 11:12 am

Hi William,

I've tried connecting different E1's and even SIP trunks but whatever I try I get a dropped call after a second. I've tested the E1 trunks on another pbx and I know they work.

[Aug 18 17:04:12] -- Executing [907843036503@default:1] AGI("SIP/101-00000000", "agi://127.0.0.1:4577/call_log") in new stack
[Aug 18 17:04:12] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Aug 18 17:04:12] -- Executing [907843036503@default:2] Dial("SIP/101-00000000", "DAHDI/g1/07843036503||To") in new stack
[Aug 18 17:04:12] -- Requested transfer capability: 0x00 - SPEECH
[Aug 18 17:04:12] -- Called g1/07843036503
[Aug 18 17:04:12] -- DAHDI/1-1 is proceeding passing it to SIP/101-00000000
[Aug 18 17:04:13] -- Channel 0/1, span 1 got hangup request, cause 28
[Aug 18 17:04:13] WARNING[3456]: app_dial.c:801 wait_for_answer: Unable to forward voice or dtmf
[Aug 18 17:04:13] -- Hungup 'DAHDI/1-1'
[Aug 18 17:04:13] == Everyone is busy/congested at this time (1:0/0/1)
[Aug 18 17:04:13] -- Executing [907843036503@default:3] Hangup("SIP/101-00000000", "") in new stack
[Aug 18 17:04:13] == Spawn extension (default, 907843036503, 3) exited non-zero on 'SIP/101-00000000'
[Aug 18 17:04:13] -- Executing [h@default:1] DeadAGI("SIP/101-00000000", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----28-----CHANUNAVAIL----------") in new stack
[Aug 18 17:04:13] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0

I'm think I've got the extensions.conf setup correct:

TRUNKX=DAHDI/g1

exten => _9XXXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9XXXXXXXXXXX,2,Dial(${TRUNKX}/${EXTEN:1},,To)
exten => _9XXXXXXXXXXX,3,Hangup

Could the settings I'm entering during wancfg_dahdi be incorrect. I've tried lots of different options none have been successful so far. Here are my dahdi_scan results for my one active port:

[1]
active=yes
alarms=OK
description=wanpipe1 card 0
name=WPE1/0
manufacturer=Sangoma Technologies
devicetype=A104
location=SLOT=1, BUS=1
basechan=1
totchans=31
irq=28
type=digital-E1
syncsrc=0
lbo=0 db (CSU)/0-133 feet (DSX-1)
coding_opts=HDB3
framing_opts=CCS,CRC4
coding=HDB3
framing=CCS/CRC4

by **williamconley** » Thu Aug 18, 2011 11:34 am

perhaps you should step away from "vicidial" and work on "asterisk" (much larger audience). after your hardware works with asterisk, vicidial will use it. the manufacturer usually has support for asterisk so you don't send the equip back.

by **will_brook** » Fri Aug 19, 2011 2:26 am

Hi William,

Thanks for the advice I'll contact Sangoma and see if they can assist.

Regards

Will

by **williamconley** » Fri Aug 19, 2011 10:21 am

once you have it working in asterisk, bring your functional extensions.conf to us and we'll help you rewrite it for Vicidial.

And don't forget to back up before you move on. LOL (backup script in /usr/share/astguiclient ... execute it with "--help" for details, stores backups in /var/log/astguiclient/archive).

by **will_brook** » Wed Aug 24, 2011 4:30 am

Thanks for your help we've managed to resolve the issue now. It was a dial plan issue. Our telco confirmed that they were seeing 12 digit numbers coming through with a 9 in front (instead of the 11 we'd normally expect). So our dial plan was incorrect, I'm guessing that the EXTEN:1 was actually just removing the _ and our Telco doesn't require a 9 to be passed to them:

exten => _9XXXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9XXXXXXXXXXX,2,Dial(${TRUNKX}/${EXTEN:1},,To)
exten => _9XXXXXXXXXXX,3,Hangup

One of my colelagues figured it out by changing the dial plan to:

exten => _XXXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _XXXXXXXXXXX,2,Dial(${TRUNKX}/${EXTEN:1},,tTor)
exten => _XXXXXXXXXXX,3,Hangup

We also changed the dial options to tTor, but this wouldn't of fixed the issue.

I took your advise and contacted Sangoma support who were very helpful and took a packet capture that enabled our Telco to see they were dropping the calls.

by **williamconley** » Wed Aug 24, 2011 8:21 pm

Actually _ isn't a number. So I'm a bit stumped there. But interesting.

The _ is only an identifier stating "this is a pattern". It does not represent a digit or character or anything that is passed.

vicidial.org

OUtbound dialing using analog Sangoma A400 card Dahdi

OUtbound dialing using analog Sangoma A400 card Dahdi

sangoma a400 configuration

sangoma a400

vicidial outbound calls

Sangoma card configuration on Vicidial

Sangoma card configuration on Vicidial

Still not adding the 9 after changes applied

still can't call

still can't call

I'll contact Sangoma

problem resolved

Who is online